1. Technical Field of the Invention
This invention relates to telecommunication systems and, more particularly, to a system and method of minimizing the number of voice transcodings during a conference call in a packet-switched network, such as a Voice-over-IP (VoIP) network, in which Tandem Free Operation (TFO) is utilized to control transcodings of the speech signal to reduce transmission cost and improve speech quality.
2. Description of Related Art
During a call between two Mobile Stations (i.e., an MS-to-MS call) in today""s circuit-switched Time Division Multiple Access (TDMA) cellular system, the speech signal is encoded with Algebraic Code Excited Linear Prediction (ACELP) encoding within the originating MS for transmission over the air interface. The signal is decoded to the Pulse Code Modulation (PCM) format within the first associated transcoder, which is typically located in the originating serving Mobile Switching Center (MSC). The PCM signals are then transported within the fixed part of the network to the terminating serving MSC utilizing a G.711 link operating at 64 kbps. A second transcoder in the terminating serving MSC converts the signal back to ACELP for transmission over the air interface to the terminating MS. Thus, there are two codecs in tandem operation in the call path. This degrades the speech quality. Additionally, at 64 kbps, the G.711 transmission requires more bandwidth.
Tandem Free Operation (TFO) is an existing methodology utilized to improve speech quality by using only one transcoder in an MS-to-MS speech path, and to reduce the long distance transmission cost by transmitting ACELP signals all the way through the backbone network. TFO is applicable only to MS-to-MS calls, however, and does not support calls from an MS to a H.323 client. For calls from an MS to a PSTN telephone, TFO utilizes a single speech coding mode (such as ACELP) from the MS all the way to the gateway connected to the PSTN network.
TFO therefore avoids repetitious coding and decoding, and in an MS-to-MS call, transports the encoded ACELP speech all the way from the originating MS to the terminating MS. However, a problem arises when a call has been set up with TFO, and a three-way conference call is then initiated. If MS1 calls MS2, and the conversation is using ACELP all the way from MS1 to MS2, TFO cannot be maintained if MS3 joins in a conference call. The call has to fall back to the G.711 mode in the link from the originating serving MSC1 to the originating gateway (GW1) connecting the call to the packet-switched network. This is because the Conference Call Device (CCD) which bridges the conference call in MSC1 is not compatible with ACELP, and can only operate with G.711 at 64 kbps. Therefore, a second transcoding must be performed in GW1 to convert the voice signal back to ACELP for transport to MS2. This setup exists today, and is only a slight problem since two transcodings do not excessively degrade the speech quality.
The problem becomes more obvious when the first call is established between MS1 and a wireline telephone in the PSTN. ACELP is used between MS1 and the terminating gateway (GW4) where it is transcoded to G.711 because the PSTN cannot operate with ACELP. If a third party mobile station (MS3) joins in a conference call, there is still the requirement for MSC1 to fall back to G.711 because of the limitation of the CCD. Once again, this does not cause a serious problem between MS1 and MS3 because there is still only two transcodings. However, there is a much greater negative impact on voice quality between S1 and the PSTN telephone because there are three transcodings required: (1) ACELP-to-G.711 in MSC1; (2) G.711-to-ACELP in GW1; and (3) ACELP-to-G.711 in GW4.
There are no known prior art teachings of a solution to the aforementioned deficiency and shortcoming. In order to overcome these deficiencies and shortcomings, it would be advantageous to have a system and method of minimizing the number of voice transcodings during a conference call in a VoIP network in which TFO is utilized to control transcodings of the speech signal. The present invention provides such a system and method.
In one aspect, the present invention is a method of minimizing the number of transcodings of a speech signal during a conference call between a first mobile subscriber, a second subscriber, and a third subscriber when the call is transported over a packet-switched network, such as a Voice-over-IP (VoIP) network, in which Tandem Free Operation (TFO) is utilized to control transcoding of the speech signal. The method begins by establishing a first call between the first mobile subscriber and the second subscriber using TFO. A gateway connecting the second subscriber to the network (second gateway) sends a message to a gateway connecting the first subscriber to the network (first gateway). The message indicates the speech coding mode being utilized between the second subscriber and the second gateway. When a Mobile Switching Center (MSC) serving the first subscriber receives an input indicating that the third subscriber should be joined in the call, communications are established between the first gateway and a third gateway connecting the third subscriber to the network. This is followed by sending a message from the third gateway to the first gateway indicating the speech coding mode being utilized between the third subscriber and the third gateway. A Conference Call Device (CCD) in the MSC establishes a three-way call bridge which provides a first call path to the first subscriber, a second call path to the second subscriber, and a third call path to the third subscriber. The first call path, utilizing a speech coding mode supported by the CCD, goes to the transcoder located in the MSC. The first call path leg between the transcoder and the MS then utilize""s speech coding supported by the MS. The second and third call paths go to the gateway connecting the first subscriber and utilize a speech coding mode supported by the CCD. The gateway connecting the first subscriber encodes the speech signal for the call leg to the second subscriber with the speech coding mode being utilized between the second subscriber and the second gateway. Likewise, the first gateway encodes the speech signal for the call leg to the third subscriber with the speech coding mode being utilized between the third subscriber and the third gateway.
In another aspect, the present invention is a system for minimizing the number of transcodings of a speech signal during a VoIP conference call in a packet-switched network which TFO is utilized. The system includes means for establishing a first call between the first mobile subscriber and the second subscriber using TFO, a first gateway connecting the first mobile subscriber to the network, a second gateway connecting the second subscriber, and a third gateway connecting the third subscriber. The second gateway includes means for sending a message to the first gateway indicating a speech coding mode being utilized between the second gateway and the second subscriber. The third gateway includes means for sending a message to the first gateway indicating a speech coding mode being utilized between the third gateway and the third subscriber. Upon initiation of a conference call joining the third subscriber, the system establishes communications between the first gateway and the third gateway. An MSC serving the first mobile subscriber includes a CCD for establishing a three-way call bridge. The call bridge provides a first call path to the first subscriber, a second call path to the second subscriber, and a third call path to the third subscriber. The second and third call paths go to the first gateway and use a speech coding mode supported by the CCD. The first gateway includes a first speech signal encoder for encoding the speech signal for the call leg to the second subscriber with the speech coding mode being utilized between the second gateway and the second subscriber. The first gateway also includes a second speech signal encoder for encoding the speech signal for the call leg to the third subscriber with the speech coding mode being utilized between the third gateway and the third subscriber.
In yet another aspect, the present invention is an access gateway for enabling a mobile subscriber to access a packet-switched network, such as a VoIP network, and to place a conference call between the first mobile subscriber, a second subscriber, and a third subscriber using TFO to control transcoding of a speech signal during the call. The access gateway includes a first connection to an MSC serving the first mobile subscriber which includes a CCD for establishing a call leg to the second subscriber and a call leg to the third subscriber over the connection to the access gateway. The access gateway also includes a second connection to a second gateway connecting the second subscriber to the network. The second connection receives a message from the second gateway indicating the speech coding mode being utilized between the second gateway and the second subscriber. A third connection within the access gateway receives a message from a third gateway connecting the third subscriber to the network. The message from the third gateway indicates a speech coding mode being utilized between the third gateway and the third subscriber. A first speech signal encoder within the access gateway encodes the call leg to the second subscriber with the speech coding mode being utilized between the second gateway and the second subscriber, and a second speech signal encoder within the access gateway encodes the call leg to the third subscriber with the speech coding mode being utilized between the third gateway and the third subscriber.